- Change "rfc2833" to "rtp" in sip.conf. Keeping backwards compatibility.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9294 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5b7c040..68f382e 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7747,7 +7747,7 @@ static const char *dtmfmode2str(int mode)
 {
        switch (mode) {
        case SIP_DTMF_RFC2833:
-               return "rfc2833";
+               return "rtp";
        case SIP_DTMF_INFO:
                return "info";
        case SIP_DTMF_INBAND:
@@ -11723,7 +11723,7 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
                ast_clear_flag(flags, SIP_DTMF);
                if (!strcasecmp(v->value, "inband"))
                        ast_set_flag(flags, SIP_DTMF_INBAND);
-               else if (!strcasecmp(v->value, "rfc2833"))
+               else if (!strcasecmp(v->value, "rfc2833") || !strcasecmp(v->value, "rtp"))
                        ast_set_flag(flags, SIP_DTMF_RFC2833);
                else if (!strcasecmp(v->value, "info"))
                        ast_set_flag(flags, SIP_DTMF_INFO);
@@ -12848,7 +12848,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
 }
 
 static char *synopsis_dtmfmode = "Change the dtmfmode for a SIP call";
-static char *descrip_dtmfmode = "SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call\n";
+static char *descrip_dtmfmode = "SIPDtmfMode(inband|info|rtp): Changes the dtmfmode for a SIP call\n";
 static char *app_dtmfmode = "SIPDtmfMode";
 
 static char *app_sipaddheader = "SIPAddHeader";
@@ -12871,7 +12871,7 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
        if (data)
                mode = (char *)data;
        else {
-               ast_log(LOG_WARNING, "This application requires the argument: info, inband, rfc2833\n");
+               ast_log(LOG_WARNING, "This application requires the argument: info, inband, rtp\n");
                return 0;
        }
        ast_mutex_lock(&chan->lock);
@@ -12889,10 +12889,10 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
        if (!strcasecmp(mode,"info")) {
                ast_clear_flag(p, SIP_DTMF);
                ast_set_flag(p, SIP_DTMF_INFO);
-       } else if (!strcasecmp(mode,"rfc2833")) {
+       } else if (!strcasecmp(mode, "rfc2833") || !strcasecmp(mode, "rtp")) {
                ast_clear_flag(p, SIP_DTMF);
                ast_set_flag(p, SIP_DTMF_RFC2833);
-       } else if (!strcasecmp(mode,"inband")) { 
+       } else if (!strcasecmp(mode, "inband")) { 
                ast_clear_flag(p, SIP_DTMF);
                ast_set_flag(p, SIP_DTMF_INBAND);
        } else

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 72bfcb5..d8a1893 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -96,11 +96,12 @@ srvlookup=yes                       ; Enable DNS SRV lookups on outbound calls
                                        ; of performing a "hairpin" call.
 ;usereqphone = no              ; If yes, ";user=phone" is added to uri that contains
                                ; a valid phone number
-;dtmfmode = rfc2833            ; Set default dtmfmode for sending DTMF. Default: rfc2833
+;dtmfmode = rtp                        ; Set default dtmfmode for sending DTMF. Default is rtp (according to RFC 2833)
                                ; Other options: 
-                               ; info : SIP INFO messages
-                               ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
-                               ; auto : Use rfc2833 if offered, inband otherwise
+                               ; info          SIP INFO messages
+                               ; inband        Inband audio
+                               ;               (requires 64 kbit codec -alaw, ulaw)
+                               ; auto : Use rtp if offered, inband otherwise
 
 ;compactheaders = yes          ; send compact sip headers.
 ;sipdebug = yes                        ; Turn on SIP debugging by default, from
@@ -412,7 +413,7 @@ srvlookup=yes                       ; Enable DNS SRV lookups on outbound calls
 ;subscribecontext=localextensions      ; Only allow SUBSCRIBE for local extensions
 ;language=de                   ; Use German prompts for this user 
 ;host=dynamic                  ; This peer register with us
-;dtmfmode=inband               ; Choices are inband, rfc2833, or info
+;dtmfmode=inband               ; Choices are inband, rtp, or info
 ;defaultip=192.168.0.59                ; IP used until peer registers
 ;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
 ;vmexten=voicemail      ; dialplan extension to reach mailbox 
@@ -428,7 +429,7 @@ srvlookup=yes                       ; Enable DNS SRV lookups on outbound calls
 ;context=from-sip              ; Context for incoming calls from this user
 ;secret=blahpoly
 ;host=dynamic                  ; This peer register with us
-;dtmfmode=rfc2833              ; Choices are inband, rfc2833, or info
+;dtmfmode=rtp                  ; Choices are inband, rtp, or info
 ;username=polly                        ; Username to use in INVITE until peer registers
                                ; Normally you do NOT need to set this parameter
 ;disallow=all